FIG. 1 shows a diagramatic scenario for a mobile (wireless) phone telecommunications link with a fixed (cable) phone receiver. The figure, for reasons of understanding, is split into a first, lower level which depicts the elements of the communications link, whilst the upper level depicts the processes realised by the elements and the link when a mobile station user employs his mobile station (MS). A microphone inside his mobile station converts speech signals into an analogue electrical signal, which anologue voice signal is transformed into a digital information stream at a rate of N kbits/s. The rate is determined by the particular codec supported (or the particular codec choice) on the mobile station and the compressed speech can be, for example in the case of the GSM system, at full rate, FR, (N=13 kbits/s, ETSI GSM 06.10), half rate, HR, (N=5.6 kbits/s, ETSI GSM 06.20) or enhanced full rate, EFR, (N=12.2 kbits/s, ETSI GSM 06.60). Other processes in turn change this digital signal into a high frequency analogue signal transmitted over the air. After being detected by a base station antenna, this radio signal is processed to recover the digital signal representing speech, which is transported over coaxial cables towards the speech transcoder. From this incoming Nkbits/s input, the speech transcoder derives another digital representation of the speech signal, at a rate of 64 kbits/s (the ITU-T G.711 standard used in fixed network transmission). It is routed through the mobile services switching centre (MSC) and various links and switches in the PSTN until it reaches the local switch to which the fixed cable telephone is connected. Typically, it is here that the analogue speech signal is reconstructed from the digital 64 kbits/s flow, and transported along the subscriber line until it reaches the telephone. An acoustic signal is then emitted by the loudspeaker which the telephone subscriber should recognise as the mobile user's voice.
At the mobile station subscriber's end, the signal can be seen as being transmitted as follows acoustic transmission, analogue transmission, digital transmission at 13 kbits/s (this transmission being performed in two different ways over the radio path and between the base station and the speech transcoder),and finally another digital transmission node in which the speech is represented by a 64 kbits/s signal.
A similar situation occurs in the case of a mobile station (MS1) communicating with a further mobile station (MS2), as depicted in FIG. 2. This situation is broadly similar, except that the audio signal transmitted for MS1 is coded and decoded a second time during the signal transmission from the base station to MS2. This results in double transcoding, compressed speech to 64 kbits/s PCM and 64 kbits/s PCM to compressed speech. This is known as tandem operation.
Tandeming is a well known source of degradation of the perceived quality of vocoded speech. This problem gets more critical, in general, when bit rates decrease. Digitial cellular systems suffer this degradation in mobile-to-mobile calls. Initially, such types of calls were not originally considered to be quite so numerous as has proven to be the case. Mobiles were previously developed primarily as entry points to the PSTN. Tandem free operation (TFO) has been for some years a subject of long discussion in various telecommunications standards committees. Nevertheless, not much work has been carried out on this subject. Recently, interest has grown appreciably, since voice quality is recognised to be a key point of the service provided to the increasing number of subscribers. Ideally, tandem free operation occurs wherein the compressed speech is not converted to PCM, the reduced number of signal representations should reduce transcoded delays and reduce transcoder errors.
64 kbits/s digital coding is well known an input analogue signal is sampled at a rate of 8 kHz, this operation limits the bandwidth for lower kHz. Each sample is given an integer value after the application of a logarithmic compression law known as the A Law or .mu. Law according to the region of the world . Each value is coded as an 8-bit symbol. The output rate is 8 kbytes/s, i.e. 64 kbits/s. The transcoding between the analogue signal and its digital A Law representation includes an analogue process, sampling a linear analogue-to-digital conversion of the samples giving a result of 13 bits, and finally a coding process which transforms the 13-bit samples into an 8-bit code.
The effect of the speech transmission methods used in the PSTN and even in ISDN cannot be neglected by the mobile transmission e.g. GSM, since they do not provide a true reconstruction of the original acoustic signal. In both cases, the high frequencies of the signal are filtered out. The lower part of the spectrum is distorted and this can give rise to some undesirable consequences. In the analogue case, the signals below 300 Hz are filtered out, in the digital case this band usually disappears because of the fixed telephone frequency range. The main consequences is that the signal processing differs from a mobile user to a fixed subscriber and vice versa. Typically, because in the mobile scenario the signal is first processed digitally before any distortion is introduced by the network resulting in the transmission quality is better in the mobile to fixed direction.